[asterisk-users] h323-sip: one way connection

[Asghar Mohammad]

nuFone h323 or openh323?


On Thu, Apr 25, 2013 at 9:33 PM, s m <sam.gh1986 at gmail.com> wrote:

> flavor? i do not understand what you mean. please explain more.
> thanks
>
>
> On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>
>> what flavor of h323 you are using?
>>
>>
>> On Wed, Apr 24, 2013 at 8:50 AM, s m <sam.gh1986 at gmail.com> wrote:
>>
>>> thanks Asghar,
>>> i do it, but no thing happened:(
>>> asterisk do not identify host line as ip address of the other end!!!!
>>>
>>>
>>> On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>>
>>>> try type=peer instead of friend.
>>>>
>>>>
>>>> On Tue, Apr 23, 2013 at 10:04 AM, s m <sam.gh1986 at gmail.com> wrote:
>>>>
>>>>> i know what is the exactly problem. i enable debug for h323 and it
>>>>> says:
>>>>> "could not find user by name 200 or address 192.168.0.146"
>>>>>
>>>>> when i change "peer-146" to "200" every thing is ok and i can call
>>>>> from two side. but it is not good for me because 200 is the name of
>>>>> extension and when i config asterisk systems, i don't know the name of
>>>>> extensions, therefore i should use addresses not name of extensions.
>>>>> do you know how i should define address of the other end in h323.conf
>>>>> file? i define the address by "host=192.168.0.146" but asterisk can not
>>>>> find it? why?
>>>>>
>>>>>
>>>>> On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>>>>
>>>>>> please post cli output for both calls.
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 22, 2013 at 11:32 AM, s m <sam.gh1986 at gmail.com> wrote:
>>>>>>
>>>>>>> hello everybody
>>>>>>>
>>>>>>> i want to have sip connection between two asterisk systems (145 and
>>>>>>> 146). connection from 145 to 146 is ok but i can not call from 146 to
>>>>>>> 145.
>>>>>>> this is h323.conf file in 145:
>>>>>>> [peer146]
>>>>>>> host=192.168.0.146
>>>>>>> type=friend
>>>>>>> context=from-trunk
>>>>>>>
>>>>>>>
>>>>>>> [to-146]
>>>>>>> type=peer
>>>>>>> host=192.168.0.146
>>>>>>> faststart=yes
>>>>>>> tunneling=no
>>>>>>> progress_audio=yes
>>>>>>> disallow=all
>>>>>>> allow=alaw
>>>>>>> allow=ulaw
>>>>>>>
>>>>>>> this is mu extensions.conf file in 145:
>>>>>>>
>>>>>>> [from-trunk]
>>>>>>> exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})
>>>>>>> [line-231]
>>>>>>> exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})
>>>>>>>
>>>>>>> i have this error: dropping call because extensions '100', 's' and
>>>>>>> 'i'
>>>>>>> doesn't exists in context default".
>>>>>>>
>>>>>>> if i change "peer146" to "general", every thing is ok and i can call
>>>>>>> from two side. my question is: in h323 connection, is it a MUST to
>>>>>>> have "general" context in h323.conf? if not, why i have this error
>>>>>>> and
>>>>>>> how i can solve it?
>>>>>>> thanks in advance
>>>>>>> sam
>>>>>>>
>>>>>>> --
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>>>>>>
>>>>>>
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>>>>>
>>>>>
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>>>>
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>>>
>>>
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>>
>>
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
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